Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On 25/03/2009 11:08 a.m., darwin.sol...@gmail.com wrote: > Test failed :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
Test --Mensaje original-- De: tracinet Remitente:asterisk-users-boun...@lists.digium.com Para:Asterisk Users Mailing List - Non-Commercial Discussion Responder a:Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP Enviado: 6 Mar, 2009 5:55 PM Basically, Server 1 is the main customer PBX where we have multiple customers running (each on their own virtual PBX separated by their contexts). Each customer has their own accountcode that we use to track calls for billing purposes, etc. The customer uses a SIP phone to register to Server 1 and sends calls to it. Server 1 in turn, passes the calls to Server 2 which is connected to various SIP providers and T-1's, etc. for termination to the PSTN. In the following sip configuration, calls work perfectly, except that the caller ID gets passed as the value from "fromuser" instead of the numeric value we set via the Set(CALLERID(num)=55) command. In other words, the fromuser overrides the caller ID value. If we remove the "fromuser" in the sip configuration, calls work great and caller ID is passed, BUT all calls land in the customerb context on Server 2 since that is the last SIP entry in sip.conf that has a host entry set to "192.168.0.11" which is the IP of Server 1. Server 1 (192.168.0.11) sip.conf [general] disallow = all allow = ulaw port = 5060 context = incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no dtmfmode=auto nat=yes ; Customer A Outbound SIP [customera-out] context=customera type=friend username=customera-out fromuser=customera-out secret= host=192.168.0.12 canreinvite=no accountcode=customera amaflags=billing dtmfmode=auto ; Customer A SIP Phone Account [customera101] context=customera type=friend username=customera101 secret=1234 host=dynamic canreinvite=no mailbox=...@customera nat=yes qualify=yes callerid="John Smith" <101> accountcode=customera amaflags=billing dtmfmode=rfc2833 ; Customer B Outbound SIP [customerb-out] context=customerb type=friend username=customerb-out fromuser=customerb-out secret= host=192.168.0.12 canreinvite=no accountcode=customerb amaflags=billing dtmfmode=auto ; Customer B SIP Phone Account [customerb101] context=customerb type=friend username=customerb101 secret=1234 host=dynamic canreinvite=no mailbox=...@customerb nat=yes qualify=yes Enviado desde mi móvil BlackBerry Orange. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On Sat, Mar 7, 2009 at 2:20 PM, Johann Steinwendtner
wrote:
> John Todd wrote:
> > Just a suggestion: have you tried more recent versions of Asterisk
> > with IAX2? I'm uncertain what version you're using, and if it's
> > 1.2.4, that's getting to be quite old and the problems that you
> > reference may already be solved in more recent updates.
> >
> > In addition, if you're set on SIP, there are features in newer
> > versions of Asterisk which allow you to both set and read SIP headers,
> > so you can insert values in those headers between Asterisk instances
> > which could then be used by the dialplan to split your calls apart
> > into different contexts or behaviors.
> >
> > See function "SIP_HEADER" and application "SIPAddHeader" for the most
> > recent versions of Asterisk.
> >
> > JT
> >
> >
> > On Mar 6, 2009, at 11:29 AM, tracinet wrote:
> >
> >> That stinks... We are migrating to SIP from IAX2 at the moment and
> >> running into the same exact problem. No way to control the
> >> destination context unless you use the "fromuser". Of course that
> >> is rendering Caller ID useless as you pointed out.
> >>
> >> I am still researching this though, if I find anything I will post
> >> it here...
> >>
> >>
> >> On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
> >> wrote:
> >> no
> >>
> >>
> >> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com
> >> ] On Behalf Of tracinet
> >> Sent: Friday, March 06, 2009 2:08 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
> >> SIP
> >>
> >>
> >>
> >> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
> >> wrote:
> >>
> >>
> >> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> >> assorted quality issues following the 1.2.4 upgrade.
> >>
> >> On the server that SENDS the call, I have the following in SIP.CONF:
> >>
> >> [192.168.1.2_OB]
> >> type=peer
> >> fromuser=OB
> >> host=192.168.1.2
> >>
> >> And in EXTENSIONS.CONF
> >>
> >> exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
> >>
> >>
> >> On the RECEIVING Server in SIP.CONF:
> >>
> >> [OB]
> >> type=user
> >> context=longdistance
> >>
> >>
> >> I am not using a REGISTER statement on the receiving server.
> >>
> >> My problem is that the only way I can seem to get the call delivered
> >> into the proper SIP context on the receiving box is to use the
> >> "fromuser=OB" on the sending machine. I tried using "username=OB",
> >> but
> >> then it delivers into the default context. I don't want to use
> >> "fromuser" because it overrides the callerid.
> >
>
> You should be able to solve the callerid issue by using the sendrpid and
> trustrpid prompts.
>
> Hans
>
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>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Hans,
Thanks for the tip - sendrpid and trustrpid is SO much more elegant
Works like a charm!!!
Thanks!
Pedro
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
John Todd wrote:
> Just a suggestion: have you tried more recent versions of Asterisk
> with IAX2? I'm uncertain what version you're using, and if it's
> 1.2.4, that's getting to be quite old and the problems that you
> reference may already be solved in more recent updates.
>
> In addition, if you're set on SIP, there are features in newer
> versions of Asterisk which allow you to both set and read SIP headers,
> so you can insert values in those headers between Asterisk instances
> which could then be used by the dialplan to split your calls apart
> into different contexts or behaviors.
>
> See function "SIP_HEADER" and application "SIPAddHeader" for the most
> recent versions of Asterisk.
>
> JT
>
>
> On Mar 6, 2009, at 11:29 AM, tracinet wrote:
>
>> That stinks... We are migrating to SIP from IAX2 at the moment and
>> running into the same exact problem. No way to control the
>> destination context unless you use the "fromuser". Of course that
>> is rendering Caller ID useless as you pointed out.
>>
>> I am still researching this though, if I find anything I will post
>> it here...
>>
>>
>> On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
>> wrote:
>> no
>>
>>
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com
>> ] On Behalf Of tracinet
>> Sent: Friday, March 06, 2009 2:08 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
>> SIP
>>
>>
>>
>> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
>> wrote:
>>
>>
>> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
>> assorted quality issues following the 1.2.4 upgrade.
>>
>> On the server that SENDS the call, I have the following in SIP.CONF:
>>
>> [192.168.1.2_OB]
>> type=peer
>> fromuser=OB
>> host=192.168.1.2
>>
>> And in EXTENSIONS.CONF
>>
>> exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
>>
>>
>> On the RECEIVING Server in SIP.CONF:
>>
>> [OB]
>> type=user
>> context=longdistance
>>
>>
>> I am not using a REGISTER statement on the receiving server.
>>
>> My problem is that the only way I can seem to get the call delivered
>> into the proper SIP context on the receiving box is to use the
>> "fromuser=OB" on the sending machine. I tried using "username=OB",
>> but
>> then it delivers into the default context. I don't want to use
>> "fromuser" because it overrides the callerid.
>
You should be able to solve the callerid issue by using the sendrpid and
trustrpid prompts.
Hans
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
the function "SIP_HEADER" and application "SIPAddHeader" seems to work
nicely upon initial testing... Thanks for the tip! Out of curiosity, are
they any standards in additional header names for the caller ID values I am
trying to add to the headers? I am using "X-CID" for now...
Thanks again!
On Fri, Mar 6, 2009 at 4:23 PM, John Todd wrote:
>
> Just a suggestion: have you tried more recent versions of Asterisk
> with IAX2? I'm uncertain what version you're using, and if it's
> 1.2.4, that's getting to be quite old and the problems that you
> reference may already be solved in more recent updates.
>
> In addition, if you're set on SIP, there are features in newer
> versions of Asterisk which allow you to both set and read SIP headers,
> so you can insert values in those headers between Asterisk instances
> which could then be used by the dialplan to split your calls apart
> into different contexts or behaviors.
>
> See function "SIP_HEADER" and application "SIPAddHeader" for the most
> recent versions of Asterisk.
>
> JT
>
>
> On Mar 6, 2009, at 11:29 AM, tracinet wrote:
>
> > That stinks... We are migrating to SIP from IAX2 at the moment and
> > running into the same exact problem. No way to control the
> > destination context unless you use the "fromuser". Of course that
> > is rendering Caller ID useless as you pointed out.
> >
> > I am still researching this though, if I find anything I will post
> > it here...
> >
> >
> > On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
> > wrote:
> > no
> >
> >
> > From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com
> > ] On Behalf Of tracinet
> > Sent: Friday, March 06, 2009 2:08 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
> > SIP
> >
> >
> >
> > On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
> > wrote:
> >
> >
> > I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> > assorted quality issues following the 1.2.4 upgrade.
> >
> > On the server that SENDS the call, I have the following in SIP.CONF:
> >
> > [192.168.1.2_OB]
> > type=peer
> > fromuser=OB
> > host=192.168.1.2
> >
> > And in EXTENSIONS.CONF
> >
> > exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
> >
> >
> > On the RECEIVING Server in SIP.CONF:
> >
> > [OB]
> > type=user
> > context=longdistance
> >
> >
> > I am not using a REGISTER statement on the receiving server.
> >
> > My problem is that the only way I can seem to get the call delivered
> > into the proper SIP context on the receiving box is to use the
> > "fromuser=OB" on the sending machine. I tried using "username=OB",
> > but
> > then it delivers into the default context. I don't want to use
> > "fromuser" because it overrides the callerid.
> >
> > Any suggestions?
> >
> > Thanks,
> > Adam
> >
> > The contents of this email message and any attachments are
> > confidential and are intended solely for addressee. The information
> > may also be legally privileged. This transmission is sent in trust,
> > for the sole purpose of delivery to the intended recipient. If you
> > have received this transmission in error, any use, reproduction or
> > dissemination of this transmission is strictly prohibited. If you
> > are not the intended recipient, please immediately notify the sender
> > by reply email and delete this message and its attachments, if any.
> >
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > Did you ever get a resolution on this?
> >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
I am on Asterisk 1.4.23.1. What you propose is interesting. I will look
into this ASAP to see if this will help. Thanks!
On Fri, Mar 6, 2009 at 4:23 PM, John Todd wrote:
>
> Just a suggestion: have you tried more recent versions of Asterisk
> with IAX2? I'm uncertain what version you're using, and if it's
> 1.2.4, that's getting to be quite old and the problems that you
> reference may already be solved in more recent updates.
>
> In addition, if you're set on SIP, there are features in newer
> versions of Asterisk which allow you to both set and read SIP headers,
> so you can insert values in those headers between Asterisk instances
> which could then be used by the dialplan to split your calls apart
> into different contexts or behaviors.
>
> See function "SIP_HEADER" and application "SIPAddHeader" for the most
> recent versions of Asterisk.
>
> JT
>
>
> On Mar 6, 2009, at 11:29 AM, tracinet wrote:
>
> > That stinks... We are migrating to SIP from IAX2 at the moment and
> > running into the same exact problem. No way to control the
> > destination context unless you use the "fromuser". Of course that
> > is rendering Caller ID useless as you pointed out.
> >
> > I am still researching this though, if I find anything I will post
> > it here...
> >
> >
> > On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
> > wrote:
> > no
> >
> >
> > From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com
> > ] On Behalf Of tracinet
> > Sent: Friday, March 06, 2009 2:08 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
> > SIP
> >
> >
> >
> > On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
> > wrote:
> >
> >
> > I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> > assorted quality issues following the 1.2.4 upgrade.
> >
> > On the server that SENDS the call, I have the following in SIP.CONF:
> >
> > [192.168.1.2_OB]
> > type=peer
> > fromuser=OB
> > host=192.168.1.2
> >
> > And in EXTENSIONS.CONF
> >
> > exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
> >
> >
> > On the RECEIVING Server in SIP.CONF:
> >
> > [OB]
> > type=user
> > context=longdistance
> >
> >
> > I am not using a REGISTER statement on the receiving server.
> >
> > My problem is that the only way I can seem to get the call delivered
> > into the proper SIP context on the receiving box is to use the
> > "fromuser=OB" on the sending machine. I tried using "username=OB",
> > but
> > then it delivers into the default context. I don't want to use
> > "fromuser" because it overrides the callerid.
> >
> > Any suggestions?
> >
> > Thanks,
> > Adam
> >
> > The contents of this email message and any attachments are
> > confidential and are intended solely for addressee. The information
> > may also be legally privileged. This transmission is sent in trust,
> > for the sole purpose of delivery to the intended recipient. If you
> > have received this transmission in error, any use, reproduction or
> > dissemination of this transmission is strictly prohibited. If you
> > are not the intended recipient, please immediately notify the sender
> > by reply email and delete this message and its attachments, if any.
> >
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > Did you ever get a resolution on this?
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ---
> John Todd
> email:jt...@digium.com
> Digium, Inc. | Asterisk Open Source Community Director
> 445 Jan Davis Drive NW - Huntsville AL 35806 - USA
> direct: +1-256-428-6083 http://www.digium.com/
>
>
>
>
> ___
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>
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
Basically, Server 1 is the main customer PBX where we have multiple customers running (each on their own virtual PBX separated by their contexts). Each customer has their own accountcode that we use to track calls for billing purposes, etc. The customer uses a SIP phone to register to Server 1 and sends calls to it. Server 1 in turn, passes the calls to Server 2 which is connected to various SIP providers and T-1's, etc. for termination to the PSTN. In the following sip configuration, calls work perfectly, except that the caller ID gets passed as the value from "fromuser" instead of the numeric value we set via the Set(CALLERID(num)=55) command. In other words, the fromuser overrides the caller ID value. If we remove the "fromuser" in the sip configuration, calls work great and caller ID is passed, BUT all calls land in the customerb context on Server 2 since that is the last SIP entry in sip.conf that has a host entry set to "192.168.0.11" which is the IP of Server 1. Server 1 (192.168.0.11) sip.conf [general] disallow = all allow = ulaw port = 5060 context = incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no dtmfmode=auto nat=yes ; Customer A Outbound SIP [customera-out] context=customera type=friend username=customera-out fromuser=customera-out secret= host=192.168.0.12 canreinvite=no accountcode=customera amaflags=billing dtmfmode=auto ; Customer A SIP Phone Account [customera101] context=customera type=friend username=customera101 secret=1234 host=dynamic canreinvite=no mailbox=...@customera nat=yes qualify=yes callerid="John Smith" <101> accountcode=customera amaflags=billing dtmfmode=rfc2833 ; Customer B Outbound SIP [customerb-out] context=customerb type=friend username=customerb-out fromuser=customerb-out secret= host=192.168.0.12 canreinvite=no accountcode=customerb amaflags=billing dtmfmode=auto ; Customer B SIP Phone Account [customerb101] context=customerb type=friend username=customerb101 secret=1234 host=dynamic canreinvite=no mailbox=...@customerb nat=yes qualify=yes callerid="Jane Jones" <101> accountcode=customerb amaflags=billing dtmfmode=rfc2833 Server 2 (192.168.0.12) sip.conf: [general] disallow = all allow=ulaw port = 5060 context = incoming canreinvite=no nat=no dtmfmode=auto [customera-out] context=customera type=friend username=customera-out secret= host=192.168.0.11 accountcode=customera amaflags=billing dtmfmode=auto [customerb-out] context=customerb type=friend username=customerb-out secret= host=192.168.0.11 accountcode=customerb amaflags=billing dtmfmode=auto On Fri, Mar 6, 2009 at 2:48 PM, Steve Howes wrote: > > On 6 Mar 2009, at 19:29, tracinet wrote: > > > That stinks... We are migrating to SIP from IAX2 at the moment and > > running into the same exact problem. No way to control the > > destination context unless you use the "fromuser". Of course that > > is rendering Caller ID useless as you pointed out. > > Give me the exact sip.conf you have both ends. Might be able to get it > working. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that you
reference may already be solved in more recent updates.
In addition, if you're set on SIP, there are features in newer
versions of Asterisk which allow you to both set and read SIP headers,
so you can insert values in those headers between Asterisk instances
which could then be used by the dialplan to split your calls apart
into different contexts or behaviors.
See function "SIP_HEADER" and application "SIPAddHeader" for the most
recent versions of Asterisk.
JT
On Mar 6, 2009, at 11:29 AM, tracinet wrote:
> That stinks... We are migrating to SIP from IAX2 at the moment and
> running into the same exact problem. No way to control the
> destination context unless you use the "fromuser". Of course that
> is rendering Caller ID useless as you pointed out.
>
> I am still researching this though, if I find anything I will post
> it here...
>
>
> On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins
> wrote:
> no
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com
> ] On Behalf Of tracinet
> Sent: Friday, March 06, 2009 2:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using
> SIP
>
>
>
> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
> wrote:
>
>
> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> assorted quality issues following the 1.2.4 upgrade.
>
> On the server that SENDS the call, I have the following in SIP.CONF:
>
> [192.168.1.2_OB]
> type=peer
> fromuser=OB
> host=192.168.1.2
>
> And in EXTENSIONS.CONF
>
> exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
>
>
> On the RECEIVING Server in SIP.CONF:
>
> [OB]
> type=user
> context=longdistance
>
>
> I am not using a REGISTER statement on the receiving server.
>
> My problem is that the only way I can seem to get the call delivered
> into the proper SIP context on the receiving box is to use the
> "fromuser=OB" on the sending machine. I tried using "username=OB",
> but
> then it delivers into the default context. I don't want to use
> "fromuser" because it overrides the callerid.
>
> Any suggestions?
>
> Thanks,
> Adam
>
> The contents of this email message and any attachments are
> confidential and are intended solely for addressee. The information
> may also be legally privileged. This transmission is sent in trust,
> for the sole purpose of delivery to the intended recipient. If you
> have received this transmission in error, any use, reproduction or
> dissemination of this transmission is strictly prohibited. If you
> are not the intended recipient, please immediately notify the sender
> by reply email and delete this message and its attachments, if any.
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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> Did you ever get a resolution on this?
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---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On 6 Mar 2009, at 19:29, tracinet wrote: > That stinks... We are migrating to SIP from IAX2 at the moment and > running into the same exact problem. No way to control the > destination context unless you use the "fromuser". Of course that > is rendering Caller ID useless as you pointed out. Give me the exact sip.conf you have both ends. Might be able to get it working. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
That stinks... We are migrating to SIP from IAX2 at the moment and running
into the same exact problem. No way to control the destination context
unless you use the "fromuser". Of course that is rendering Caller ID
useless as you pointed out.
I am still researching this though, if I find anything I will post it
here...
On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins wrote:
> no
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *tracinet
> *Sent:* Friday, March 06, 2009 2:08 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
>
>
>
>
>
> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
> wrote:
>
>
> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> assorted quality issues following the 1.2.4 upgrade.
>
> On the server that SENDS the call, I have the following in SIP.CONF:
>
> [192.168.1.2_OB]
> type=peer
> fromuser=OB
> host=192.168.1.2
>
> And in EXTENSIONS.CONF
>
> exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
>
>
> On the RECEIVING Server in SIP.CONF:
>
> [OB]
> type=user
> context=longdistance
>
>
> I am not using a REGISTER statement on the receiving server.
>
> My problem is that the only way I can seem to get the call delivered
> into the proper SIP context on the receiving box is to use the
> "fromuser=OB" on the sending machine. I tried using "username=OB", but
> then it delivers into the default context. I don't want to use
> "fromuser" because it overrides the callerid.
>
> Any suggestions?
>
> Thanks,
> Adam
>
> The contents of this email message and any attachments are confidential and
> are intended solely for addressee. The information may also be legally
> privileged. This transmission is sent in trust, for the sole purpose of
> delivery to the intended recipient. If you have received this transmission
> in error, any use, reproduction or dissemination of this transmission is
> strictly prohibited. If you are not the intended recipient, please
> immediately notify the sender by reply email and delete this message and its
> attachments, if any.
>
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> Did you ever get a resolution on this?
>
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
On the RECEIVING Server in SIP.CONF:
[OB]
type=user
context=longdistance
I am not using a REGISTER statement on the receiving server.
My problem is that the only way I can seem to get the call delivered
into the proper SIP context on the receiving box is to use the
"fromuser=OB" on the sending machine. I tried using "username=OB", but
then it delivers into the default context. I don't want to use
"fromuser" because it overrides the callerid.
Any suggestions?
Thanks,
Adam
The contents of this email message and any attachments are confidential
and are intended solely for addressee. The information may also be
legally privileged. This transmission is sent in trust, for the sole
purpose of delivery to the intended recipient. If you have received this
transmission in error, any use, reproduction or dissemination of this
transmission is strictly prohibited. If you are not the intended
recipient, please immediately notify the sender by reply email and
delete this message and its attachments, if any.
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Did you ever get a resolution on this?
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins wrote:
>
> I am switching from IAX2 to SIP for my inter-Asterisk transport due to
> assorted quality issues following the 1.2.4 upgrade.
>
> On the server that SENDS the call, I have the following in SIP.CONF:
>
> [192.168.1.2_OB]
> type=peer
> fromuser=OB
> host=192.168.1.2
>
> And in EXTENSIONS.CONF
>
> exten => 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
>
>
> On the RECEIVING Server in SIP.CONF:
>
> [OB]
> type=user
> context=longdistance
>
>
> I am not using a REGISTER statement on the receiving server.
>
> My problem is that the only way I can seem to get the call delivered
> into the proper SIP context on the receiving box is to use the
> "fromuser=OB" on the sending machine. I tried using "username=OB", but
> then it delivers into the default context. I don't want to use
> "fromuser" because it overrides the callerid.
>
> Any suggestions?
>
> Thanks,
> Adam
>
> The contents of this email message and any attachments are confidential and
> are intended solely for addressee. The information may also be legally
> privileged. This transmission is sent in trust, for the sole purpose of
> delivery to the intended recipient. If you have received this transmission
> in error, any use, reproduction or dissemination of this transmission is
> strictly prohibited. If you are not the intended recipient, please
> immediately notify the sender by reply email and delete this message and its
> attachments, if any.
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Did you ever get a resolution on this?
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